Transmission

Transm ission This document runs through some transmission basics:

An alo gue Connectivi ty Li ne Types This uses mainly PSTN dialup lines of varying quality and modems to provide digital connections. Circuit switching is used to provide a link for a particular call and levels of quality line types can be offered by the telephone companies e.g. 1. Line Type 1 - Basic voice, 2. Line Type 5 - Basic data, 3. Line Type 7 - voice and data over private lines. In addition, you can obtain conditioned lines with improved communications. There is D Cond iti oning and levels 1 to 8 of C Cond it ioning . A more expensive dedicated analogue line can be bought where the circuit is fixed and is not different every time that you dial up, thereby giving you a more consistent service.

Circu it S wit ch ed P ath When making an analogue phone call you first obtain a dial tone, then you dial a number. This number is sent to the local switch containing a D-Channel Bank using touch tone Dual Tone Mult ifr equency (D TMF) signals. That is as far as DTMF gets. The voice call is converted to the digital Pulse Code Modulation format and analogue signalling to digital signalling by the D-channel bank. The switch routes the call from this point through the digital switch network using the Mana gement (M)-Plane protocol called Signa ll ing System Number 7 (S S7) which is a form of CCS. SS7 sends messages to the switch which is connected to the destination phone and this far end switch sends a Cont rol (C)-Plane message that rings the far end phone. When the phone is picked up

the C-Plane mechanisms send the message that the path is available. The digitised voice is the User (U )-Plane data flow.

Digit al Signal ling DS0 The Modern Telco digitizes speech using Pulse Code Mo dulat ion (PCM) on 64K (DS0) channels. 64 Kbps is considered to be Digi tal Signa l L ev el 0. Each channel is sampled 8000 times/second according to Nyquist's Theorem, and incorporates 8 bits per sample (hence 8bits x 8000 giving 64,000 bits/sec). This figure of 8000 comes from the fact that the valuable range of telephone signals is 100Hz to 4kHz, and the sampling rate is twice that of the highest signal. The standard G.711 defines the Pulse Code Mo dulat ion (PCM) 64Kb/s voice channel. DS0 trunks make up the trunks around the digital network that can carry data or voice. For voice the conversion to 2-wire analogue occurs at the switch closest to the user. The call handling in the 'external' network is dealt with by SS7 . Straight digital signals (bipolar) are used across these lines so no modem is required. A Channel Servi ce U nit/Data Servi ce Unit (CS U/DSU) provides the interface for the end user and converts the DTE's digital signals into the Synchronous digital signals used over the WAN. Kilostream services (BT's version of Digi tal Data Servi ces ) offer from 2.4Kb/s to 64Kb/s (56Kb/s in the USA) whilst Kilostream N allows multiples of between 2 times and 16 times 64Kb/s to give a range of bandwidths from 128Kb/s to 1024Kb/s. BT's Megastream offers 2, 8, 34, 45, 140, 155Mb/s bandwidths for really bandwidth intensive traffic.

T1 (DS -1)

Fr aming For the DS-1, also called T1, Time Division Mu lti ple xing (TDM) is used to transport multiple channels over one line. Clocking of the serial transmission needs to occur at one end of the link or the other, sometimes you will see the clocking options as interna l i.e. provided by the local device, or line meaning

that the clock is provided by the remote device. Two-pairs are used in a T1 link. The T1 link can operate in full-duplex mode where one pair transmits and the other pair receives. 24 channels are available for transmission and these are grouped together to form a Fr ame i.e. the 24 time slots (8 bits each) plus one framing bit form one T1 frame (193 bits, the 193rd bit being the synchronisation/framing bit). For 8000 samples a second, a T1 frame must be transmitted every 125 usecs, we can therefore calculate the T1 line rate as 193 x 8000 = 1.544 Mbps (A DS0 line rate is 8 bits x 8000 = 64 Kbps). The frames can also be grouped into 12 sequenced frames to form a Super frame (S F) (also called a D4 ) which means that 12 framing bits are used per SF. These 12 framing bits are also called F bits. They form the sequence 10 00 11 01 11 00 and are used to sequence the SF within 4 frames. In one second 8000 'F' bits are used for framing. This is encapsulated in the G.704 framing standard. A D4 contains 288 channels. The frames could also be grouped into 24 to form the newer framing format called the Exten ded Super frame (ES F) . The 8000 'F' bits are used differently in ESF where 2000 'F' bits are used for framing, 2000 are used for CRC-6 error checking and 4000 are used as a supervisory channel for things such as loopback and error reporting. An ESF contains 576 channels. Channel As soc iated Signa ll ing (CAS) T1 signalling can take the form of CAS using Robbed Bit Signa ll ing where bits are 'robbed' from the channels carrying the voice. This is called In-ban d Signa ll ing . In the SF, the LSB is 'robbed' from each of the 24 x 8-bit timeslots in the 6th and the 12th frames. The A bit comes from the 6th frame timeslots whereas the B bit comes from the 12th frame timeslots. These 'robbed' bits are used for call supervision and trunk signalling in the voice environment e.g. the 'A' bit is commonly used in the same way that the 'M' lead is used in E&M signalling i.e. signalling by pulsing the 'A' bit. This Bit Robbing is fine if the channels are used for voice because the 8 bit samples being reduced to 7 bits every 6 frames does not significantly impact on voice quality. Data is of course not so forgiving with the lowered quality line so each channel is reduced to 56kbps for data (In the US a type of ISDN called Swit ched Servi ces uses bit-robbing technology that results in a 56kbps B-channel). The problem with using CAS is that these robbed bits are really only used when setting up and establishing a call, the rest of the time the bandwidth is wasted. The only messages used are Wink , Ringin g, Hang up and Pulse Dig it Dia ll ing .

The ESF operates a similar manner to the SF other than bits are robbed from the 18th frame (C bits ) and the 24th frame (D bits ). The main difference between channelised lines (analogue) and non-channelised lines (ISDN) is that they do not have a built-in D-channel. For example, all 24 channels on a T1 line only carry data. The signalling is in-band or associated to the data channels (Channel A ssociat ed Signal lin g (C AS) ). Traditional channelised lines do not support digitized data calls (for example, BRI with 2B+D). Channelised lines support a variety of in-band signal types, such as ground start, loop start, wink start, immediate start, E&M and R2. Common Chan ne l Signal ling (CCS) T1 signalling can also take the form of CCS which is normally Common Channel Signa ll ing Number 7 (S S7) or Primary Rate ISDN where one channel (D-channel, channel 24) is used for Q.931 signalling. This is called Out of -band Signa ll ing since the signalling is in a channel that is separate from the voice channels. This speeds up call setup by up to a factor of 5, to 1-3 seconds. One signalling channel can handle up to 1500 calls. SS7 is a protocol in its own right, very akin to X.25 where switches exchange billing, switching and signalling information. With CCS, PRI does not operate Bit Robbing but takes one of the channels and uses that for signalling (D-channel) instead leaving 23 channels for the data. The line encoding coding scheme used to allow both data and voice is ususally based on a pseudo-ternary bipolar code called Bipolar wi th 8 -Zer os Substitut ion (B 8ZS) . This is called Clear Channel . Another coding scheme called B7 exists for voice only applications and yet another called Al ternate Mark Inversion (AMI) is commonly used.

E1

Fr aming The European standard E1 is another way of using TDM to transport multiple channels over a single line. The E1 interface has 32 channels or time slots. Framing is carried out in time slot 17, the 32 x 8-bit slots are grouped into a 256 bit Fr ame and then 16 frames are grouped into a Mul ti frame . Time slot 17

(channel 16) in the first frame of the Multiframe indicates the beginning of the Multiframe. In the first frame there are the 4 signalling bits (ABCD) for channel 1 and 4 bits (ABCD) for channel 17. In the second frame there are 4 signalling bits (ABCD) for channel 2 and 4 bits (ABCD) for channel 18. This carries on 16 times to form the Multiframe. G.704 covers this framing for E1 (as well as T1). The E1 digital line operates at 2048Kb/s and is made up of 32 x 64Kb/s. Often, the E1 is broken up into 64Kb/s channels and is known as Fr actional E1 (or FT1 in the States) if the customer is presented with a group, for example a 384Kb/s circuit (made up of 6 x 64Kb/s channels). In this instance, the carrier determines the grouping of the channels, one E1 may server a number of different clients. A channelised T1/E1 (CT1/CE1) line is an analogue line that was originally intended to support analogue voice calls, but has evolved to support analogue data calls. ISDN does not transmit across channelised T1/E1 lines. Al ternate Mark Inversion (AMI) can be used with E1 but the most common E1 line-encoding scheme used is called Hi gh-Densit y Bipo lar with 3- zer os (HDB3) . The error checking used is called Cy clic Redundancy Check with lev el 4 checking (CR C-4 ), although this can be turned off (no-CRC4) with some providers. Australia has a different way of E1 framing from the rest of the world. Channel As soc iated Signa ll ing (CAS) In CAS, Time slot 17 (E0 channel 16) is used for signalling and time slot 1 (E0 channel 0) is used for framing synchronisation and alarms, the other 30 are used for voice and data. The CAS signalling is considered in-band and is very simple as it just identifies four states: • • • •

00 01 10 11

-

Idle Seizure Disconnect Busy

Common Chan ne l Signal ling (CCS)

E1 can also be set up for CCS where channel 16 carries signalling such as Q.931 for Primary Rate ISDN (I.421). This signalling, based on HDLC type protocols can often be vendor proprietory signalling that needs to be transparently passed by network equipment. Examples include BT's Dig ita l Priv ate Network Signa ll ing System (DPNS S) , Nortel's Mer idian Customer Def ined Netwo rking (MC DN) , QSIG and Signa ll ing System 7 (S S7) which is a standard for CO to CO signalling used throughout the world.

Sig nal ling Band widt hs One channel can carry a different data signal from another and therefore allows multiplexing to occur to give 32 simultaneous data transmissions. The 64Kb/s data rate is known as Digi tal Signa l L ev el 0 (DS-0) and the 1.544Mb/s rate is known as DS-1 (T1), the following table details some of the transmission rates: Signal Type

Carrie r

No . of T1 channels

DS-0

No . of V oice channels

Data R ate (Mb/s

1

0.064

DS-1

T1

1

24

1.544

DS-1C

T1-C

2

48

3.152

DS-2

T2

4

96

6.312

DS-3

T3

28

672

44.736

DS-4

T4

168

4032

274.760

Multiplexers combines several channels into a single bit stream. Synch ronous data transmission differs from Asynchronous transmission in that data is sent as a continuous stream of data packets separated by start and stop bits conforming to precise clocking which is governed either by one end of a link, or the other. Asynchronous transmission has no central precise clocking and packets can come and go at undefined times and in different orders with different frequencies. Each end of an Asynchronous link is responsible for its own clocking. For interest sake, Isynch ronous transmission is when asynchronous data is sent over synchronous transmission. Plesio chr onous transmission is when digital signals with different, but reliable, clock signals are being used.

Up until this point we have looked at Circui t S witched Netw orks , however, there are also Pack et Swit ched Networks where data is split into individual packets and each packet is switched separately through the WAN. These means that packets can end up at the destination in a different order from when they left. For this reason, each packet is tagged with a destination address and order number. Often called An y-to- An y connect ions , Packet switched networks are only switching small packets, so resends are fast and the whole switching process is a fast one. Virtual circuits are set up to provide either a temporary path Switche d Virtual Ci rcu it (S VC) or a Permanent path Permanent Virtual Circui t (P VC) .

QS IG The QSIG CCS protocol is significant because it is based on Q.931 and Q.933 and is being developed to cover the first three layers of the OSI model with layer 3 being the messaging layer. The messaging facilities are more complex allowing for features such as Call Forwarding, CTI and ACD to be extended across the digital trunks between different PBX manufacturers.

Sign al ling S ys tem 7 (S S7) SS7 is a form of CCS called CCS number 7 and has been defined by the ITU as ITU -T#7 . SS7 uses the signalling channel on T1 (24) or E1 (17) to carry signalling, billing and switching information between SS7 capable switches. This is carried out-of-band in parallel to the calls.

Terminal s and Mo dems Various types of Terminal access over the Wide Area network use different methods to mimic being directly connected to the host: • •

Telnet/ rlo gin - provides a virtual terminal connection to a computer. The protocol rlogin is specific to Unix systems. Local- area T ran spo rt ( LAT) - is provided by DEC to allow terminals to connect to a number of hosts at one time.

• •

TCP/IP T elnet 3 27 0 (TN32 70) - is the virtual terminal protocol that allows one to access 3270 applications. X.2 5 P ack et As semble r/Disassembler (P AD) - The PAD translates the character-based terminal output into X.25 packets that can be switched on the Wide Area Network. X.25 has versatility due to its addressing schemes.

Dial-in Modems are used when there are Asy nch ronous connections to the Wide Area Network. Protocols that run over the asynchronous connections include Ser ial Line Pr ot oco l ( SLIP) , Point to Point Pr ot oco l (PPP) and App leTalk Remote Access Pr oto col (ARAP) . Traditionally, modems converted digital signals to analogue to traverse the local loop where they are converted to digital as they are routed across the network before being converted back into analogue for the local loop at the remote end where the modem at the remote client converted from analogue to digital again. The theoretical speed limit for data throughput using this scenario is 33kbps. With the V.90 56kbps standard the the digital-to-analogue conversion still occurs at the client end but instead of converting back to analogue at the remote end, say at an ISP, the signal remains digital and can do so provided that the service provider has a digital link to the telco. This means that although the data speed in one direction from the client to the ISP is still limited to a maximum of 33kbps, the data speed from the ISP to the client (download) can theoretically reach 56kbps. If both ends have to convert back to analogue signalling then the maximum speed in either direction will only be 33kbps, for example modem to modem connection between two private individuals. The rate between a computer and a modem can be up to 57.6 kbps for a V.32bisPlus modem or 115.2 kbps for a V.34 modem. These higher speeds are achieved using V.42bis compression (level 5 of the Mi cr ocom Networking Protoc ol (MNP) ). Modems commonly monitor the line quality and apply Adap tiv e Speed Lev el lin g ( ASL) and lower the speed if the line quality deteriorates thereby keeping the modem connection open.